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gerrit-public.fairphone.software
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platform
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external
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webrtc
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3e4c77f1c12800b1ab0eceee1b675538600e2473
3e4c77f
Fix AGC2 fixed-adaptive gain controllers order.
by Alessio Bazzica
· 6 years ago
096d016
Update MultiplexEncoderAdapter to use EncoderInfo
by Erik Språng
· 6 years ago
58df0ad
Add guards to VideoCaptureDS::Init for when pins are null
by Andreas Pehrson
· 6 years ago
9b5b070
Use EncoderInfo in SimulcastEncoderAdapter
by Erik Språng
· 6 years ago
4eb4112
Plug-in media transport state listener
by Piotr (Peter) Slatala
· 6 years ago
189013b
Update QualityTestVideoEncoder to use GetEncoderInfo()
by Erik Språng
· 6 years ago
449afd9
Updated ScopedVideoEncoder to use GetEncoderInfo()
by Erik Språng
· 6 years ago
5e78461
Make the extra seturation margin configurable.
by Alex Loiko
· 6 years ago
b1e031a
JitterEstimator: Remove old LowRate exp and add trial for upper bound.
by Erik Språng
· 6 years ago
15ca5a9
Add implicit conversion between rtc:PacketTime and int64_t.
by Niels Möller
· 6 years ago
96965ae
Add ability to enable frame dumping decoder via field trial.
by Erik Språng
· 6 years ago
fe45da4
Remove WebRTC-VP8-GfBoost field trial.
by philipel
· 6 years ago
af6d741
Makes send time information in feedback non-optional.
by Sebastian Jansson
· 6 years ago
be837ac
Remove RTPFragmentationHeader from LibvpxVp8Encoder.
by philipel
· 6 years ago
2812763
Remove deprecated AudioProcessing::GetStatistics function
by Sam Zackrisson
· 6 years ago
4e93329
Properly setup MockPeerConnectionObserver in tests (continued).
by Yves Gerey
· 6 years ago
dd20c9c
Add support for screen sharing with PipeWire on Wayland
by Tomas Popela
· 6 years ago
7f4dfa4
Remove locks from AEC2 and move it into private_submodules_
by Sam Zackrisson
· 6 years ago
59844ce
Revert "Use the factory instead of using the builtin code path in `VideoCodecInitializer`."
by Qingsi Wang
· 6 years ago
7852d29
Improve the documentation of MdnsResponderInterface and rename MDns.* to Mdns.*.
by Qingsi Wang
· 6 years ago
eb2c641
Delete the default implementations of MediaTransportInterface methods.
by Bjorn Mellem
· 6 years ago
be14217
Use the factory instead of using the builtin code path in `VideoCodecInitializer`.
by Jiawei Ou
· 6 years ago
8386435
Roll chromium_revision 6271fcdc14..734e273d43 (604273:604373)
by chromium-webrtc-autoroll
· 6 years ago
1f6aa9f
Add interfaces for using MediaTransport as the transport for data channels.
by Bjorn Mellem
· 6 years ago
062a691
Roll chromium_revision 9996ac8918..6271fcdc14 (604166:604273)
by chromium-webrtc-autoroll
· 6 years ago
9f95625
When SDES is used, pass pre-shared key to media transport.
by Piotr (Peter) Slatala
· 6 years ago
7182286
Allow FakeNetworkPipe to wake up its processing thread
by Sebastian Jansson
· 6 years ago
693432d
Add obj-c mapping from native configuration to RTCConfiguration
by Piotr (Peter) Slatala
· 6 years ago
e6caa9f
export RTCRtpTransceiverInit
by Piasy
· 6 years ago
ed45c57
Corrects audio overhead correction in Scenario test.
by Sebastian Jansson
· 6 years ago
69807e8
Depend directly on destination targets.
by Yves Gerey
· 6 years ago
a8fa2d0
Move some methods from StreamInterface to FifoBuffer
by Niels Möller
· 6 years ago
21cddff
Harmonize paths to dependent targets.
by Yves Gerey
· 6 years ago
b32bb95
Bugfix: FlexFEC causes retransmit bitrate increase.
by Ying Wang
· 6 years ago
8b7d206
AEC3: Decrease latency until the delay has been detected
by Per Åhgren
· 6 years ago
f577ab3
Roll chromium_revision 7e85c0922c..9996ac8918 (604065:604166)
by chromium-webrtc-autoroll
· 6 years ago
b00b28e
Roll chromium_revision 0cb3899c4e..7e85c0922c (603959:604065)
by chromium-webrtc-autoroll
· 6 years ago
b3f887b
Expose key derivation through a simple interface for use in WebRTC.
by Benjamin Wright
· 6 years ago
1a92cd7
Roll chromium_revision 34bb9a9162..0cb3899c4e (603839:603959)
by chromium-webrtc-autoroll
· 6 years ago
c78b0ea
Create a MediaTransportState enum and add a state callback to MediaTransport.
by Bjorn Mellem
· 6 years ago
eaf337a
Remove event wait logic from DesktopConfigurationMonitor
by Emircan Uysaler
· 6 years ago
746d46b
AGC2: renaming GainCurveApplier to Limiter.
by Alessio Bazzica
· 6 years ago
fcc3981
Revert "Use only first payload timestamp for RTCP SR generation for audio"
by Ilya Nikolaevskiy
· 6 years ago
992a868
Fix for clock reset repair.
by Christoffer Rodbro
· 6 years ago
a2e133d
Delete StreamInterface::ReadLine.
by Niels Möller
· 6 years ago
ed7b8b1
Update media transport settings struct
by Piotr (Peter) Slatala
· 6 years ago
3e67676
Add support for field trials in peerconnection_client|server
by Bjorn Terelius
· 6 years ago
9a0662a
Use only first payload timestamp for RTCP SR generation for audio
by Ilya Nikolaevskiy
· 6 years ago
b26cf2f
Add field trial to enable the new RTC event log format.
by Bjorn Terelius
· 6 years ago
97e35ce
Revert "Disabled TestPacketBuffer.SeqNumWrapOneFrame test due to clang update"
by Artem Titarenko
· 6 years ago
0eb7d3ff
Always call ConvertToI420 with positive crop_height
by Robert Bares
· 6 years ago
9862c2e
Delete OptionsFile class. Refactored only user, TurnFileAuth.
by Niels Möller
· 6 years ago
3df6e715
Makes PacketResult::GetSentPacket const.
by Sebastian Jansson
· 6 years ago
b33168e
Roll chromium_revision 89ed1da2c8..34bb9a9162 (603733:603839)
by chromium-webrtc-autoroll
· 6 years ago
946179c
Delete unused function rtc::Flow.
by Niels Möller
· 6 years ago
42b4315
Add iOS SDK unit tests for nalu_rewriter
by Artem Titarenko
· 6 years ago
9190b82
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
by Johannes Kron
· 6 years ago
f43bcd4
Remove likely obsolete entries from WATCHLISTS
by Oleh Prypin
· 6 years ago
0ac98ab
Roll chromium_revision 03b56190ff..89ed1da2c8 (603619:603733)
by chromium-webrtc-autoroll
· 6 years ago
770c32a
Roll chromium_revision 55624cc6cd..03b56190ff (603513:603619)
by chromium-webrtc-autoroll
· 6 years ago
3b149e4
Added myself to the base watchlist to monitor ssl* changes.
by Benjamin Wright
· 6 years ago
5124a04
Roll chromium_revision 62e33bd2f0..55624cc6cd (603177:603513)
by chromium-webrtc-autoroll
· 6 years ago
6b9d823
Add TargetBitrate callback to MediaTransportInterface.
by Piotr (Peter) Slatala
· 6 years ago
c640a93
Fix import of chromium into webrtc.
by Artem Titov
· 6 years ago
a0677d1
Add MediaTransportSettings struct for configuring media transport.
by Piotr (Peter) Slatala
· 6 years ago
12048c7
Fix error handling in hex_decode.
by Niels Möller
· 6 years ago
ef45669
Adds GetSentPacket to PacketResult.
by Sebastian Jansson
· 6 years ago
449c1c0
Adds unit tests for safe reset trial.
by Sebastian Jansson
· 6 years ago
7286496
Download aap2 and bundletool as part of required dependencies.
by Yves Gerey
· 6 years ago
6fcf6ca
Modified PressEnterToContinue() to actualy check if Enter is pressed
by Danail Kirov
· 6 years ago
2c16cc6
Replace some usage of EventWrapper with rtc::Event.
by Niels Möller
· 6 years ago
88d8d7d
Add missing assignment in RTCConfiguration.mm
by Piotr (Peter) Slatala
· 6 years ago
f3ff14c
Properly setup MockPeerConnectionObserver in tests.
by Yves Gerey
· 6 years ago
22a8f98
Formatted sslidenty.cc and moved non referenced functions into an
by Benjamin Wright
· 6 years ago
428320c
Formatting OpenSSLCertificate and doing some minor code cleanup.
by Benjamin Wright
· 6 years ago
5d35554
Rename private member functions to use CamelCase.
by Benjamin Wright
· 6 years ago
61c5cc8
Makes OpenSSL concrete implementations final.
by Benjamin Wright
· 6 years ago
2616594
Refactor: Make SSLCertChain a final class.
by Benjamin Wright
· 6 years ago
150a907
FrameEncryption Video End To End Testcase.
by Benjamin Wright
· 6 years ago
c462a6e
Prevent the frame decryptor being set if the channel is stopped.
by Benjamin Wright
· 6 years ago
625771d
Roll chromium_revision a539a24569..62e33bd2f0 (603045:603177)
by chromium-webrtc-autoroll
· 6 years ago
59ebf23
Refactor structs in rtc_event_log_parser_new.h
by Elad Alon
· 6 years ago
ff43541
Delta compression efficiency improvement for non-existent base
by Elad Alon
· 6 years ago
436ebca
Fix extra setdscp call introduced by bad merge.
by Tim Haloun
· 6 years ago
0f08d22
Add a function for enabling the congestion window and pushback controller in the webrtc::SendSideCongestionController.
by erikvarga@webrtc.org
· 6 years ago
99b71df
Use function_video_(en|de)coder_factory from api
by Danil Chapovalov
· 6 years ago
88c2c50
Use monotonic clock to derive NTP timestamps in RTCP module
by Ilya Nikolaevskiy
· 6 years ago
fdee701
Add parser and unittests for new RTC event log format.
by Bjorn Terelius
· 6 years ago
916ae08
Makes critsect_.Leave() more visible in PacedSender.
by Sebastian Jansson
· 6 years ago
6dd7f91
Remove deprecated deregistration functions in RtcpTransceiver
by Danil Chapovalov
· 6 years ago
06aa209
Add support to adapt video without preserving aspect ratio
by Magnus Jedvert
· 6 years ago
9049037
Simplify api/DEPS presubmit check.
by Mirko Bonadei
· 6 years ago
7d76a31
Use MediaTransportInterface, for audio streams.
by Niels Möller
· 6 years ago
2769cd5
Roll chromium_revision f54583b6a0..a539a24569 (602763:603045)
by chromium-webrtc-autoroll
· 6 years ago
0d24772
Allocate CMBlockBuffers using a memory pool.
by Kári Tristan Helgason
· 6 years ago
c35096d
Reland "Encode RTC event logs in new format."
by Bjorn Terelius
· 6 years ago
a5d543c
Set minSdkVersion to 16 for AppRTCMobile_stubbed_video_io_test_apk.
by Mirko Bonadei
· 6 years ago
95dfa52
Clarify the desired semantics of AsyncResolverInterface::GetResolvedAddress.
by Zach Stein
· 6 years ago
327b753
Split out a separate target for VP8EncoderSimulcastProxy
by Jonathan Yu
· 6 years ago
5abd541
Stop exporting simulcast_encoder_adapter.h in :rtc_internal_video_codecs
by Jonathan Yu
· 6 years ago
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